It is demanded in a mobile communication system that speech signals are compressed to low bit rates to transmit to efficiently utilize radio wave resources and so on. On the other hand, it is also demanded that quality improvement in phone call speech and call service of high fidelity be realized, and, to meet these demands, it is preferable to not only provide quality speech signals but also encode other quality signals than the speech signals, such as quality audio signals of wider bands.
The technique of integrating a plurality of coding techniques in layers is promising for these two contradictory demands. This technique combines in layers the base layer for encoding input signals in a form adequate for speech signals at low bit rates and an enhancement layer for encoding differential signals between input signals and decoded signals of the base layer in a form adequate to other signals than speech. The technique of performing layered coding in this way have characteristics of providing scalability in bit streams acquired from an encoding apparatus, that is, acquiring decoded signals from part of information of bit streams, and, therefore, is generally referred to as “scalable coding (layered coding).”
The scalable coding scheme can flexibly support communication between networks of varying bit rates thanks to its characteristics, and, consequently, is adequate for a future network environment where various networks will be integrated by the IP (Internet Protocol).
For example, Non-Patent Document 1 discloses a technique of realizing scalable coding using the technique that is standardized by MPEG-4 (Moving Picture Experts Group phase-4). This technique uses CELP (Code Excited Linear Prediction) coding adequate to speech signals, in the base layer, and uses transform coding such as AAC (Advanced Audio Coder) and TwinVQ (Transform Domain Weighted Interleave Vector Quantization) with respect to residual signals subtracting base layer decoded signal from original signal, in the enhancement layer.
Further, to flexibly support a network environment in which transmission speed dynamically fluctuates due to handover between different types of networks and the occurrence of congestion, scalable encoding of small bit rate scales needs to be realized and, accordingly, needs to be configured by providing multiple layers of lower bit rates.
Patent Document 1 and Patent Document 2 disclose a technique of transform encoding of transforming a signal which is the target to be encoded, in the frequency domain and encoding the resulting frequency domain signal. In such transform encoding, first, an energy component of a frequency domain signal, that is, gain (i.e. scale factor) is calculated and quantized on a per subband basis, and a fine component of the above frequency domain signal, that is, shape vector, is calculated and quantized.    Non-Patent Document 1: “All about MPEG-4,” written and edited by Sukeichi MIKI, the first edition, Kogyo Chosakai Publishing,    Patent Document 1: Japanese Translation of PCT Application Laid-Open No. 2006-513457    Patent Document 2: Japanese Patent Application Laid-Open No. HEI7-261800